Jean Cérien | 16 May 2013 16:31
Picon

multiple wan proxying

 
Hello All !
 
I am trying to setup an asterisk server with multiple nated nic and multiple dsl links (one nic = one dsl  link = one ip) in order to allow clients to use indiferently a nic/public ip (to register and exchange media)
 
I am thinking of using siproxd to achieve this. I'd like siproxd to listen on a given interface (say eth1), possibly with a specific port if it helps, and that it redirects all this traffic to asterisk on port 5060 / eth0 - reciprocically, asterisk would reply on port /eth1 (or lo0) and siproxd would push all this to the client via eth1 and its public adress
 
Is this feasible ?
 
note: I've installed siproxd 0.8.1 - but no default config file - does it have to be created by hand ?
 
Thanks to all
 
J.
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Jack Howard | 16 Apr 2013 13:00
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Compilation issue - checking for osip_init in -losip2... no - *** ERROR: libosip2 is required!

Hi!


I'm trying to compile siproxd on a Fedora-based Linux system.

I've downloaded and compiled libosip2, but when I try and run the ./configure for siproxd, it errors:

# ./configure --with-libosip-prefix=/usr/local
...
...
checking for osip_init in -losip2... no
*** ERROR: libosip2 is required!   Maybe you need to use --with-libosip-prefix ?

However, as far as I can see everything is where it should be.  I've tried setting the prefix to /usr/local and /usr/local/lib, and my /etc/ld.so.conf contains both paths too:

include ld.so.conf.d/*.conf
/usr/local/lib
/usr/lib/mysql
/usr/local/mysql-5.0.67-linux-i686/lib
/usr/local

An objdump appears to show the required item present:

# objdump -x /usr/local/lib/libosip2.so |grep osip_init
00008fa6 g     F .text  0000015a              osip_init

Have tried Googling, which lead me to the use of objdump, but no further.  

All help appreciated.

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Jeremy Malcolm | 7 Apr 2013 09:54
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Configuration help needed for Cisco 7905

My configuration is like this one:

http://siproxd.sourceforge.net/siproxd_guide/siproxd_guide_c7s3.html

However my phone is different; it's a Cisco 7905.  The relevant settings that the phone allows me to
configure are these:

UID	 cisco
PWD	 mypassword
Proxy	 my.sip.server.net
SIPRegOn     1
SIPRegInterval	 3600
StaticIP    192.168.15.122
OutBoundProxy	 192.168.15.200 [my siproxd machine]
StaticRoute    192.168.15.1 [my dumb masquerading router, with siproxd set as the DMZ]

Here is my configuration for siproxd:

if_inbound  = eth0
if_outbound = eth0
host_outbound = name-that-resolves-to-my-external-ip.net

It doesn't work, here is what happens:

15:31:35 INFO:siproxd.c:233 siproxd-0.8.1-53 i486-pc-linux-gnu starting up
15:31:35 INFO:plugins.c:112 Plugin 'plugin_logcall' [Logs calls to syslog] loaded with success, exemask=0x40
07:31:35 INFO:rtpproxy_relay.c:121 Current thread stacksize is 8192 kB
07:31:35 INFO:sock.c:131 bound to port 5060
07:31:35 INFO:siproxd.c:344 siproxd-0.8.1-53 i486-pc-linux-gnu started
07:31:57 INFO:plugin_logcall.c:126 Outgoing Call:
cisco@... ->
888123456@... [Req: 888123456@...]
07:31:57 INFO:plugin_logcall.c:126 ACK Call:
cisco@... ->
888123456@... [Req: 888123456@...]

Also by using ngrep I see this:

interface: eth0 (192.168.15.0/255.255.255.0)
filter: (ip or ip6) and ( port 5060 )
#
U 192.168.15.122:5060 -> 192.168.15.200:5060
REGISTER sip:my.sip.server.net SIP/2.0.
Via: SIP/2.0/UDP 192.168.15.122:5060.
From: sip:cisco@...;tag=3840067541.
To: sip:cisco@...
Call-ID: 4035595390@...
CSeq: 1 REGISTER.
Contact: Jeremy Malcolm <sip:cisco@...:5060;transport=udp>;expires=3600.
User-Agent: Cisco-CP7905/1.01-030807A.
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER.
Content-Length: 0.
.

#
U 192.168.15.200:5060 -> 192.168.15.122:5060
SIP/2.0 408 Request Timeout.
Via: SIP/2.0/UDP 192.168.15.122:5060.
From: <sip:cisco@...>;tag=3840067541.
To: <sip:cisco@...>.
Call-ID: 4035595390@...
CSeq: 1 REGISTER.
Content-Length: 0.
.

#
U 192.168.15.122:5060 -> 192.168.15.200:5060
INVITE sip:88812346@...;user=phone SIP/2.0.
Via: SIP/2.0/UDP 192.168.15.122:5060.
From: Jeremy Malcolm <sip:cisco@...>;tag=816758886.
To: <sip:88812346@...;user=phone>.
Call-ID: 2441701698@...
CSeq: 1 INVITE.
Contact: Jeremy Malcolm <sip:cisco@...:5060;transport=udp>.
User-Agent: Cisco-CP7905/1.01-030807A.
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER.
Expires: 300.
Content-Length: 262.
Content-Type: application/sdp.
.
v=0.
o=cisco 15826 15826 IN IP4 192.168.15.122.
s=Cisco 7905 SIP Call.
c=IN IP4 192.168.15.122.
t=0 0.
m=audio 16384 RTP/AVP 18 8 0 101.
a=rtpmap:18 G729/8000/1.
a=rtpmap:8 PCMA/8000/1.
a=rtpmap:0 PCMU/8000/1.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.

#
U 192.168.15.200:5060 -> 192.168.15.122:5060
SIP/2.0 408 Request Timeout.
Via: SIP/2.0/UDP 192.168.15.122:5060.
From: Jeremy Malcolm <sip:cisco@...>;tag=816758886.
To: <sip:88812346@...;user=phone>.
Call-ID: 2441701698@...
CSeq: 1 INVITE.
Content-Length: 0.
.

#
U 192.168.15.122:5060 -> 192.168.15.200:5060
ACK sip:88812346@...;user=phone SIP/2.0.
Via: SIP/2.0/UDP 192.168.15.122:5060.
From: Jeremy Malcolm <sip:cisco@...>;tag=816758886.
To: <sip:88812346@...;user=phone>.
Call-ID: 2441701698@...
CSeq: 1 ACK.
User-Agent: Cisco-CP7905/1.01-030807A.
Content-Length: 0.

Interestingly there is no sign of "name-that-resolves-to-my-external-ip.net" above.  I have tried
running a similar command on my.sip.server.net (because it's my own SIP server running yate), but it
never sees anything.

Also interestingly, I have sip softphones on the same network which work fine, even without using siproxd
as the proxy.  I guess they are a bit smarter about NAT, and can work fine even though my router does not have a
sip ALG built-in.

Any help appreciated...

PS. Public IP addresses/domains have been changed for privacy/security reasons.

--
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Internet and Open Source lawyer, consumer advocate and geek
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Yves Gancberg | 21 Dec 2012 12:03
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Siproxd Pfsense help...

Hello Siprox Gurus,

I have a little question for you:
I am trying to use siproxd on a Pfsense box, so that a SIP Server on the LAN interface (LAN !) can act as a SIP server for phones on the WAN (so, the other way around, as we normally do)...

The Sip server therefore has a private addresse (192.168.2.100).
I would like siproxd to act as an ALG (Application Layer Gateway), so that it modifies the SIP packets (content of the application layer) in a way that it matches the UDP packets (so that my packets are matching at the layer 4 and 7)...

So far, I was unable to do so, I and I unable to find any log / debug information... I am even not sure siproxd is simply 'running' on my pfsense...

Who could give me a hand / redirect me to a decent 'howto' manual ?

Thanks in advance for your answer,

Yves.

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Henning Holtschneider | 25 Sep 2012 19:09
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Splitting of comma-separated headers

Hello,

I'm using siproxd 0.8.1 on Debian Squeeze. I have a problem with comma-separated header values being split
into multiple headers. For example, the header

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO

becomes

Allow: INVITE
Allow: ACK
Allow: CANCEL
Allow: OPTIONS
Allow: BYE
Allow: REFER
Allow: SUBSCRIBE
Allow: NOTIFY
Allow: INFO

after going through siproxd. This behaviour is completely valid, but it results in significantly larger
packets. Sometimes, the packet size is beyond the MTU on my internal network and the packets become
fragmented. Is it possible to change the behaviour of siproxd so that headers with comma-separated
values remain unchanged?

Regards,
Henning Holtschneider
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Maximilian Ruta | 5 Sep 2012 21:19

Using siproxd to open a local sip server

Hi,

i want to use siproxd to open my Asterisk server to the public. The Asterisk server is behind a NAT and i cannot
forward ports to it. But the Asterisk server is in an VPN together with an server that has an public ip address.

So this is how i want my setup to be:

 Asterisk  
  as SIP    --- OpenVPN --- Proxy Server
 registrar                       |
                             Internet
                                 |
                             SIP Clients

Is this posible with siproxd? I try to setup it with this configuration:

[root <at> jupiter ~]# ifconfig 
eth0: flags=4163<UP,BROADCAST,RUNNING,MULTICAST>  mtu 1500  metric 1
        inet 94.249.134.17  netmask 255.255.255.0  broadcast 94.249.134.255
        inet6 fe80::4042:8dff:feb0:2f42  prefixlen 64  scopeid 0x20<link>
        ether 42:42:8d:b0:2f:42  txqueuelen 1000  (Ethernet)
        RX packets 26757603  bytes 3029780057 (2.8 GiB)
        RX errors 0  dropped 32  overruns 0  frame 0
        TX packets 18925600  bytes 47581287242 (44.3 GiB)
        TX errors 0  dropped 0 overruns 0  carrier 0  collisions 0

lo: flags=73<UP,LOOPBACK,RUNNING>  mtu 16436  metric 1
        inet 127.0.0.1  netmask 255.0.0.0
        inet6 ::1  prefixlen 128  scopeid 0x10<host>
        loop  txqueuelen 0  (Local Loopback)
        RX packets 1234740  bytes 186405699 (177.7 MiB)
        RX errors 0  dropped 0  overruns 0  frame 0
        TX packets 1234740  bytes 186405699 (177.7 MiB)
        TX errors 0  dropped 0 overruns 0  carrier 0  collisions 0

tap0: flags=4163<UP,BROADCAST,RUNNING,MULTICAST>  mtu 1500  metric 1
        inet 10.8.0.1  netmask 255.255.0.0  broadcast 10.8.255.255
        inet6 fe80::9810:f8ff:fe3c:67c6  prefixlen 64  scopeid 0x20<link>
        ether 9a:10:f8:3c:67:c6  txqueuelen 100  (Ethernet)
        RX packets 800165  bytes 165082677 (157.4 MiB)
        RX errors 0  dropped 0  overruns 0  frame 0
        TX packets 930041  bytes 150851885 (143.8 MiB)
        TX errors 0  dropped 0 overruns 0  carrier 0  collisions 0

[root <at> jupiter ~]# cat /etc/siproxd.conf
if_inbound  = eth0
if_outbound = tap0
host_outbound = 10.8.100.1
sip_listen_port = 5060
daemonize = 0
silence_log = 1
registration_file = /var/lib/siproxd/siproxd_registrations
autosave_registrations = 300
pid_file = /var/run/siproxd/siproxd.pid
rtp_proxy_enable = 1
rtp_port_low  = 7070
rtp_port_high = 7089
rtp_timeout = 300
rtp_dscp = 46
sip_dscp = 0
rtp_input_dejitter  = 0
rtp_output_dejitter = 0
tcp_timeout = 600
tcp_connect_timeout = 500
tcp_keepalive = 20
debug_level =      0x00000000
debug_port = 0
ua_string = Siproxd-UA
plugindir=/usr/lib/siproxd/
load_plugin=plugin_logcall.la
load_plugin=plugin_fix_bogus_via.la

The output is in the attachment try1.log.

I tryed it also with this configuration:

[root <at> jupiter ~]# cat /etc/siproxd.conf
if_inbound  = eth0
if_outbound = tap0
sip_listen_port = 5060
daemonize = 0
silence_log = 1
registration_file = /var/lib/siproxd/siproxd_registrations
autosave_registrations = 300
pid_file = /var/run/siproxd/siproxd.pid
rtp_proxy_enable = 1
rtp_port_low  = 7070
rtp_port_high = 7089
rtp_timeout = 300
rtp_dscp = 46
sip_dscp = 0
rtp_input_dejitter  = 0
rtp_output_dejitter = 0
tcp_timeout = 600
tcp_connect_timeout = 500
tcp_keepalive = 20
debug_level =      0x00000000
debug_port = 0
ua_string = Siproxd-UA
plugindir=/usr/lib/siproxd/
load_plugin=plugin_logcall.la
load_plugin=plugin_fix_bogus_via.la

The output is in the attachment try2.log.

Thanks

Maximilian Ruta
Attachment (try2.log): text/x-log, 132 KiB
Attachment (try1.log): text/x-log, 93 KiB
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Shannon Ward | 27 Mar 2012 17:04
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Transparent proxy of multiple clients

I'm currently testing Siproxd as a transparent proxy of multiple clients.
All outbound calls are working great.
Inbound calls intermittently have one way voice.
If you hang up and then call right back it will work fine.

Does Siproxd support transparent proxing of multiple clients or are 
there known issues
with multiple clients?

siproxd-0.8.1-53 x86_64-redhat-linux-gnu
The clients are Mobile Broadband Routers running Astrisk.

More detail can be given if needed.

Thanks
Shannon

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Reza Ambler | 21 Mar 2012 01:12
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Config Help

Hello everyone,

I’m trying to use siproxd to test proxying some phones back to our switch and have siproxd handle the media relay. I have my example configuration file here, http://sandbox.vintalk.com/~reza/work/0XDEADBEEF/siproxd.conf but it doesn’t appear to be relaying the SIP registration to my other softswitch. I was wondering if someone could help out here, I think it’s something minor I’m missing. This is my first go at using siproxd.

 

Debug Log File : http://sandbox.vintalk.com/~reza/work/0XDEADBEEF/debug.log

Thanks,

 

Reza Ambler                             
Systems Engineer

(O) 858.357.8770
(F) 858.357.8694
(E) 
reza.ambler <at> vintalk.com
(W) www.vintalk.com

 

Vintalk
9707 Waples Street, Suite 201         
San Diego, CA92121 

 

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Mark Purcell | 23 Jan 2012 22:21
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Bug#657070: embedded library ltdl

Package: siproxd
Version: 1:0.8.1-1
Severity: important
Tags: security upstream help

siproxd currently ships an embedded copy of the ltdl library.

The original version of ltdl shipped was vunerable to 
'CVE-2009-3736 local privlege esclation'

siproxd upstream (Thomas) have now upgraded the embedded copy of ltdl
as a result siproxd is no longer vunerable to CVE-2009-3736.

The current version of siproxd in Debian Fixed in version siproxd/1:0.8.1-1.

However this Debian version is still using the embedded ltdl library, rather
than the preferred system provided ltdl library.

Whilst siproxd does detect if a system ltdl library is available it does
fail to build from source (FTBFS) with the error:

plugins.c:65: undefined reference to 'lt__PROGRAM__LTX_preloaded_symbols'

which has also been reported here:
http://blog.gmane.org/gmane.network.siproxd/month=20110201

Assistance to fix this issue in the Debian package would be appreciated.

Mark

-- System Information:
Debian Release: wheezy/sid
  APT prefers unstable
  APT policy: (500, 'unstable'), (1, 'experimental')
Architecture: amd64 (x86_64)

Kernel: Linux 3.2.0-1-amd64 (SMP w/2 CPU cores)
Locale: LANG=en_AU.UTF-8, LC_CTYPE=en_AU.UTF-8 (charmap=UTF-8)
Shell: /bin/sh linked to /bin/dash

Versions of packages siproxd depends on:
ii  adduser     3.113
ii  libc6       2.13-24
ii  libosip2-7  3.6.0-2

siproxd recommends no packages.

Versions of packages siproxd suggests:
ii  linphone  3.5.0-2

-- no debconf information

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Siproxd doesnt' listen on Ipv6

Hello,

I notice that siproxd is only listening on the ipv4 adressess of my interface. That's what netstat tells me.
I cannot find any documentation about how to make siproxd work with ipv6. Is it possible?

I need to test the Ipv6 support of my SIP client.

Att.

Antonio Carlos

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Doug Holcombe | 7 Nov 2011 15:10
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siproxd errors in system logs on pfsense 2.0

Hello  

I would like to see if I can get help for this issue here. I had installed pfsense 1.2.3 along with siproxd package almost a year ago. Everything has been working just fine with my Asterisk sip connections back to callcentric.com and voip.ms. I then had a smart idea and decided to upgrade to pfsense 2.0. The upgrade went just fine until I noticed that my telephone like coming in was not hitting my asterisk box all the time. I logged into callcentric and see that it would show periodically "Your phone is not registered". I didn't have this problem before I upgraded! So in my research to resolve this I find the following errors in the system logs for pfsense. Nov 7 01:24:52 siproxd[53118]: sock.c:469 ERROR:tcp_connect() failed Nov 7 01:25:00 siproxd[53118]: proxy.c:706 ERROR:proxy_response: list_get via failed Nov 7 01:25:16 siproxd[53118]: proxy.c:706 ERROR:proxy_response: list_get via failed Nov 7 01:25:20 siproxd[53118]: proxy.c:706 ERROR:proxy_response: list_get via failed Nov 7 01:25:46 siproxd[53118]: sock.c:469 ERROR:tcp_connect() failed Nov 7 01:25:56 siproxd[53118]: sock.c:469 ERROR:tcp_connect() failed Nov 7 01:26:41 siproxd[53118]: proxy.c:706 ERROR:proxy_response: list_get via failed Nov 7 01:27:00 siproxd[53118]: proxy.c:706 ERROR:proxy_response: list_get via failed Nov 7 01:27:07 siproxd[53118]: sock.c:469 ERROR:tcp_connect() failed Nov 7 01:28:03 siproxd[53118]: sock.c:310 WARNING:recv() returned error [Connection reset by peer], disconnecting TCP [192.168.123.200] fd=13 Nov 7 01:28:26 siproxd[53118]: proxy.c:706 ERROR:proxy_response: list_get via failed Nov 7 01:28:30 siproxd[53118]: proxy.c:706 ERROR:proxy_response: list_get via failed So I figure that something is broken for this package in the new version of pfsense 2.0. I did search on the forums here to see if I can find any resolution for this but came up empty. Maybe someone could help me out as I need to have my phones working Thanks Doug --
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